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Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and objectives
    2. Categories of Voice over Internet Protocol (VoIP) and technological progression
    3. Core principles of the Session Initiation Protocol (SIP)
    4. SIP standardization framework (RFC 3261 and associated standards)
  2. System Architecture
    1. User Agent (UA) components
    2. Essential server types: Registrar, Location, Proxy, and Redirect
    3. Application server functions
    4. Entity identification and addressing schemes
    5. SIP protocol topology
  3. Server Operations
    1. User registration processes
    2. Proxy and Redirect server functionalities
    3. Stateless versus stateful proxy mechanisms
    4. Location services
    5. SRV records and Domain Name System (DNS) integration
    6. URI, URL, URN formats; ENUM services; and NAPTR records
  4. SIP Signalling Mechanisms (including Instant Messaging & Presence – IMP extensions)
    1. Message architecture
    2. Request types
    3. Response codes
    4. Illustrative call procedure
    5. Headers and parameters
    6. IMP operational models
  5. Session Description Protocol (SDP)
    1. Media description formats
    2. Standardized codec inventory
    3. Session negotiation protocols
  6. Signalling Flows – SIP
    1. SIP session establishment – primary RFC 3261 example
    2. Representative call scenarios
    3. Conferencing and IP Private Branch Exchange (PBX) integration
    4. Dynamic media modification during sessions
    5. Utilization of IMP services
  7. Routing of SIP Requests and Responses
    1. VIA header processing
    2. ROUTE and RECORD-ROUTE headers
  8. SIP to Public Switched Telephone Network (PSTN) Interworking
    1. SIP-T and SIP-I standards
    2. Early media and SIP trunking capabilities
    3. SIP-PSTN signalling integration
  9. Security Considerations in SIP
    1. Securing SIP, RTP, and RTCP traffic
    2. Common implementations for Secure SIP
  10. Operational Challenges and Future Outlook
    1. NAT and firewall traversal techniques
    2. Quality of Service (QoS) implementation
    3. SIP and SDP integration within 3GPP IMS architecture
  11. Summary and Discussion

Part II: Practical Application

  1. Deploying SIP in Local Area Network Environments: XLite SIP UA + Asterisk
    1. Establishing Asterisk accounts with basic dial plan configuration
    2. Configuring XLite SIP User Agent (DTMF, codecs, NAT, RTP, timer, registration) and SIP endpoints (Polycom, Gigaset, Yealink, Linphone)
    3. Managing registration, call initiation, and reception
    4. Peer-to-peer calling with Linphone
  2. Analyzing SIP Signalling via Wireshark
    1. Server configuration protocols
    2. Capture of SIP signalling and RTP media streams
    3. SIP packet analysis and isolation of specific call records
    4. Troubleshooting voice quality issues: Jitter buffer management, DTMF retrieval (RFC 2833, INFO), codec/DNMF diagnostics (transcoding, GSM failures, tone duplication)
    5. VoIP monitoring tools
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attributes
    2. SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practical IM&P exercises using XLite and Linphone
  4. Examining SIP Call Flows
    1. SIP Registration via DNS
      • SIP SRV record mechanics
      • Endpoint registration utilizing DNS-SRV
    2. Call Flow Analysis with DNS
      • Analyzing SIP signalling traces in Wireshark
      • Troubleshooting DNS timeouts and latency issues
  5. SIP Trunking
    1. Establishment of test SIP trunks
    2. Troubleshooting for Denial of Service (DoS), Distributed Denial of Service (DDoS), fraud, and calls per second (CPS) limits
  6. SIP Security Measures
    1. Implementing SIP security with IPsec
    2. Securing SIP communications
    3. Risks associated with IP telephony fraud
    4. Strategies for preventing DDoS and other attack vectors
  7. Deploying SIP-Based VoIP Services
    1. Switch configuration procedures
    2. SIP client setup and registration protocols
    3. Software Infrastructure
      • Asterisk PBX / FreeSWITCH softswitch / Cisco Call Manager
      • Linux CentOS environment
      • TDM-to-IP drivers
      • Softphone applications (XLite, Linphone)
    4. Hardware Components
      • Server hardware
      • TDM-to-IP cards/gateways
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone and Hardphone Management
    1. Configuration Settings
      • Codec selection
      • User credentials, SIP server/proxy details, and port assignments
    2. Operational Signalling for:
      • Three-way calling
      • Call forwarding mechanisms
      • Attendant call transfers
      • Message Waiting Indication (MWI) and Busy Lamp Field (BLF)
    3. Yealink automated provisioning
    4. Vendor-specific limitations
  9. SIP and Network Address Translation (NAT) Challenges
    1. NAT types and structural components
    2. STUN (Simple Traversal of UDP Through NATs) protocol
  10. Troubleshooting VoIP Call Quality
    1. Diagnostics for connected calls with missing media
    2. Critical QoS parameters
      • Latency, jitter, and playback buffer size
    3. VoIP quality metrics
      • RTCP analysis for delay and jitter
      • Mean Opinion Score (MOS) per ITU-T G.107 E-model
    4. Quality monitoring tools (e.g., Voipmonitor)
  11. Cloud-Based IP Telephony Solutions
  12. Summary and resolution of participant-submitted SIP and VoIP issues
 21 Hours

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